When a network element (usually a router) cannot forward packets to the next network element fast enough then the packets will get queued in that router’s internet buffer, increasing the latency (and the jitter) and potentially end up dropping packets.

When that happens and there is that increase in latency and packet loss it is when we say that the network is congested.

WebRTC endpoints try to prevent congestion by estimating the available bandwidth and limiting the bitrate sent based on the feedback received from receiver side.